1. Field of the Invention
The present invention relates to speech transmission and reception system for digital communication of audio signals.
2. Description of the Related Art
Recently, digitalization of mobile communication systems such as mobile telephones and cordless telephone is expanding rapidly. Particularly in mobile communication systems, multiplexing techniques, high efficiency speech encoding techniques, multivalue modulation and demodulation techniques and other techniques have enabled more efficient usage of a given frequency band. At the same time, developments in speech encoding technology and speech decoding technology are anticipated from the viewpoint of improvement in communication quality.
The following is a general explanation of a conventional speech transmission and reception system for digital communication.
FIG. 7 is a block diagram of a conventional speech transmission and reception system for digital communication.
The illustrated speech transmission and reception system for digital communication comprises a microphone 11, an amplifier 12A, an A/D converter 13, a speech encoder 14, a frame making portion 15A, an error correction encoder 16, a modulator 17, a propagation path 18, a demodulator 19, an error correction decoder 20, a frame processing portion 15B, a speech decoder 21, a D/A converter 22, an amplifier 12B, and a speaker 23. From the microphone 11 to the modulator 17 constitute a transmitter, while from the demodulator 19 to the speaker 23 constitute a receiver.
First, the transmitter will be explained.
The microphone 11 is a converter that converts a speech into an electric audio signal.
The amplifier 12A is an amplifier that amplifies the audio signal.
The A/D converter 13 is a circuit that samples the audio signal at a sampling rate of 8,000 cycles per second, and converts each sample to an 8-bit digital signal. Therefore, this A/D converter sends a signal to the speech encoder 14 at a rate of 64 kilobits per second (Kbps).
The speech encoder 14 is a circuit having a function that estimates a pattern of the audio signal in advance utilizing a regularity in a state transition of the audio signal, and calculates a differential between the estimated pattern and an actual pattern of the input audio signal so as to output the differential. Thus, the input audio signal is compressed and encoded. This compressing and encoding method is called Adaptive Differential Pulse Code Modulation (ADPCM). Using the ADPCM method, the input audio signal can be compressed into a half-bit size, so a 64 Kbps input signal can be converted into a 32 Kbps signal before being sent to the frame making portion 15A.
The frame making portion 15A is a circuit having a function that generates and outputs a frame every time the 32 Kbps signal for 5 milliseconds is received. The frame making portion 15A receives 160 bits of audio signal during 5 milliseconds. Then, a cyclic redundancy check (CRC) code having 16 bits is added to the audio signal to produce a frame containing 176 bits that is sent to the error correction encoder 16. This CRC code is necessary for checking a bit error in the frame received in the receiver side. If there is a bit error or plural bit errors in a frame, the frame is removed as an error frame.
The error correction encoder 16 is a circuit that receives each frame having 176 bits sequentially and performs a convolution encoding frame by fame. The convolution encoding is an encoding process of sequential data as if the data were convoluted. In other words, each of the sequential data is coded not independently, but relatedly to the previous and the following data as if it were convoluted. By this method, even if a bit error is generated in a part of the data in the propagation path, the data can be restored at a high accuracy by utilizing the data convoluted in the previous and following data. The convolution-encoded frame can be decoded by the Viterbi decoding method. In the error detection by the CRC code in the above-mentioned error correction encoder 16, a frame in which a bit error was detected is removed. In contrast, the Viterbi decoder has a function of error detection as well as error correction. In the following explanation, a frame having 176 bits is expressed as, for example, 176 bits/frame, and a signal group encoded and arranged in series is expressed as a code sequence.
The 176 bits/frame signal is doubled in the bit size to 352 bits/frame by the convolution encoding performed by the error correction encoder 16.
The modulator 17 performs digital modulation of a carrier having a specific frequency with the output of the error correction encoder 16, so as to transmit the result to the propagation path 18, which can be a wireless or a wired path.
Next, the operation of the receiver is explained.
The demodulator 19 performs digital demodulation of the signal after propagating the propagation path 18, so as to send the result to the error correction decoder 20. In general, a digital signal is constituted with binary bits, each of which is one or zero. However, the output of this demodulator 19 is a multivalue signal in which one symbol is constituted with three bits and eight levels. One symbol means a bit of received digital signal. Therefore, the bit size of the output signal of the demodulator 19 is triple that of the input signal.
The error correction decoder 20 converts the multivalue signal having three bits and eight levels sent from the demodulator 19 into a binary signal while performing error correction by the Viterbi decoding method.
Accordingly, the bit size of the output signal of the error correction decoder 20 becomes one third of the input signal. The Viterbi decoder (not illustrated) of the error correction decoder 20 has a function of performing the error correction decoding of the signal that was processed with the convolution encoding in the transmitter side, as mentioned above.
The Viterbi decoded binary signal having 176 bits/frame is sent to the frame processing portion 15B.
The frame processing portion 15B performs error detection frame by frame using the 16 bits of CRC code in the 176 bits/frame signal. If an error is detected in a frame, the frame is removed. If no error is detected, the frame is decomposed and is converted into a 32 Kbps signal, which is sent to the speech decoder 21. This signal is the identical to the ADCPM signal encoded by the speech encoder 14 in the transmitter side.
The speech decoder 21 performs ADPCM inversion so as to decode the input signal into a 64 Kbps signal, which is sent to the D/A converter 22.
The D/A converter 22 converts the 64 Kbps digital signal into an analog signal, which is sent to the amplifier 12B.
The amplifier 12B amplifies the analog signal and sends the signal to the speaker 23.
The speaker 23 converts the analog signal into speech.
As explained above, a speech received by the microphone 11 is transmitted via the propagation path 18 and is received by the receiver to be outputted from the speaker 23.
However, the above-mentioned conventional art has the following problems to be solved.
In the system shown in FIG. 7, if an error is detected in a frame by the frame processing portion 15B, the frame is removed. When the frame that is a part of the speech signal is removed, a speech skip may occur or the quality of the speech may be deteriorated. Therefore, to prevent deterioration of the speech quality, the latest frame preceding the removed frame is inputted to the speech decoder 21 again to supplement the removed frame.
However, the above-mentioned process requires complicated control, which is disadvantageous.